Media Pass

Using the Media Pass feature, you can delegate audio processing to external systems, with RustPBX handling the protocol part.

Configure in the option: CallOption parameter's mediaPass: MediaPassOption in the Invite/Accept command:

Parameters

url

WebSocket service address.

inputSampleRate

Audio sample rate received from the WebSocket connection (also the RTP/WebRTC track sample rate).

Recommended settings:

• SIP calls: 16000hz
• WebRTC calls:
  • G722: 16000hz
  • Opus: 48000hz
  • Others: 8000hz

outputSampleRate

Audio sample rate sent from the WebSocket connection (also the sample rate that external services will receive) If different from inputSampleRate, resampling will be required.

packetSize

Packet size (default: 2560 bytes).

Flow

When a call-in or call-out session is established, a Media Pass Track will be created. Like other Tracks, the start and end of the Media Pass Track will trigger Track Start and Track End events respectively.

Media Pass Track will use the configured address plus parameters to request the external websocket service. Example:

ws://127.0.0.1:8080/?sample_rate=16000&packet_size=1024

• sample_rate is the configured outputSampleRate
• packet_size is the configured packetSize

After the connection is successful, Media Pass Track will start bidirectional audio forwarding, as shown: